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Sample Record

The document is a lab record notebook for the Analog and Digital Communications Lab at SR University, detailing instructions for students, a syllabus, and a list of experiments. It includes guidelines for lab conduct, a detailed syllabus covering topics like amplitude modulation, frequency modulation, and digital modulation, as well as specific experiments to be conducted. The document also provides theoretical background, apparatus required, procedures, and MATLAB code for simulating amplitude modulation and demodulation.

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0% found this document useful (0 votes)
12 views122 pages

Sample Record

The document is a lab record notebook for the Analog and Digital Communications Lab at SR University, detailing instructions for students, a syllabus, and a list of experiments. It includes guidelines for lab conduct, a detailed syllabus covering topics like amplitude modulation, frequency modulation, and digital modulation, as well as specific experiments to be conducted. The document also provides theoretical background, apparatus required, procedures, and MATLAB code for simulating amplitude modulation and demodulation.

Uploaded by

2405a41l55
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

SCHOOL OF ENGINEERING
SR UNIVERSITY
WARANGAL

RECORD NOTEBOOK

Analog and Digital Communications


Lab
(II [Link] II Sem)

Name of the student:


_________CH.KEERTHI_______________________
Year/Semester/Section:__2/2/C-
SECTION____________________________
[Link]:_____2205A41164_____________________________
_________

Signature of Lab Incharge Signature of HOD


INSTRUCTIONS TO THE STUDENTS

Do’s

1. The students should come to the lab in time.


2. They should maintain discipline and decorum during the lab hours.
3. They should wear their ID Cards.
4. They are required to come prepared for the experiments to be conducted.
5. They should bring the observation book and submit the lab record immediately for evaluation.
6. Take the components as per the requirements of the experiment.
7. See that the equipment’s/ components are returned after completion of experiment to the concerned
person.
8. Switch off the power supply as soon on the experiment is completed.
Don’ts
1. Do not damage the IC’s/ Components.
2. Do not switch on the supply without permission of the lab supervisor.
3. Do not talk during the lab session.
4. Do not walk around the lab.
5. Do not displace the apparatus after completion of experiment.
6. Mobile phones are not permitted into the lab.
7. Do not attempt to repair the apparatus or apply excessive voltage to the equipment’s.
8. Do not carry eatables inside the laboratory.
Title Analog & Digital Communication Pre-requisite Signal & System
Code 20EC108 Credit 4
Type Program core Format (L-T-P) 3-0-2

DETAILED SYLLABUS

MODULE I (8Hr)
Introduction: Block diagram of communication systems, Need of Modulation.
Amplitude modulation (AM) and De-modulation: Definition, time and frequency domain representation, power
calculations; generation and detection of AM, DSB-SC, SSB-SC; VSB (Qualitative Information); FDM.

MODULE II (8Hr)
Frequency modulation (FM) and De-modulation: Definition and expression, Carson’s rule, generation and
detection of FM, Introduction to Phase modulation (PM).
Receivers: Characteristics of receiver, Tuned radio receiver (TRF), Super-Heterodyne receiver (SHR) of AM and
FM.

MODULE III (6Hr)

Pulse Analog Modulation: Line codes, PAM, PWM, PPM (Qualitative Information); TDM.

MODULE IV (6Hr)
Pulse Digital Modulation: Block diagram of digital communication system, generation and detection of PCM, DM,
and Adaptive DM.

MODULE V (8Hr)
Binary modulations: Generation and detection of ASK, PSK, FSK, QPSK; Error probability of digital modulations
scheme.

TEXTBOOKS/LEARNING RESOURCES:

1. Bhooshan, S. Fundamentals of Analogue and Digital Communication Systems. Singapore: Springer Nature
Singapore. 2021.
2. Bhagyaveni, M. A., Kalidoss, R., Vishvaksenan, K. S. Introduction to Analog and Digital
Communication. Denmark: River Publishers. 2022.

REFERENCE BOOKS/LEARNING RESOURCES:

1. Otung, I. Communication Engineering Principles. United Kingdom: Wiley. 2021.


2. Chen, K. Principles of Communication: A First Course in Communication. Denmark: River Publishers.
2023.

LIST OF EXPERIMENTS: (Note: Minimum 12 experiments to be conducted)


All these experiments are to be simulated first either using Commsim, MATLAB, SCILAB, OCTAVE, or any other
simulation package and then to be realized in hardware.

1. Amplitude modulation and demodulation.


2. DSB-SC Modulator & Detector
3. Frequency Modulation and Demodulation
4. Pre-emphasis and De-emphasis.
5. Pulse Amplitude Modulation & DeModulation
6. Pulse Position Modulation and Demodulation.
7. PCM generation and detection.
8. Differential pulse code modulation.
9. Delta modulation.
10. Time-division multiplexing of four bandlimited signals.
11. Generation and detection of ASK.
12. Generation and detection of FSK.
13. Generation and detection of PSK.
14. QPSK: generation and detection.
15. Study of spectrum analyzer and analysis of AM and FM signals

INDEX SHEET
[Link] Name of the Experiment Page Date of Date of Assessment Signature
Performance Submission
No Marks of Faculty
1 AM modulation and
demodulation.

2 DSBSC modulation and


demodulation.

3 FM modulation and
demodulation.

4 PAM modulation and


demodulation.

5 Time division multiplexing


and demultiplexing.

6 Generation and detection of


FSK.

7 Generation and detection of


PSK.

8 Generation and detection of


ASK.

9 Verification of sampling
theorem.

10 Verification of pulse code


modulation.
11

12
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment Amplitude modulation and demodulation


Name of the candidate [Link]
Register Number 2205a41164
Date of Experiment

Experiment-1
AMPLITUDE MODULATION AND DEMODULATION
Objective:

1. To understand the operation of amplitude modulation and demodulation and to calculate depth of modulation for
various modulating voltages.
2. To simulate Amplitude modulation and demodulation.

Apparatus Required:
□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit
□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
1. CRO with probes □Patch cards □RPS
□Micro phone □Speakers

THEORY:
In order that a steady radio signal or "radio carrier" can carry information it must be changed or modulated in one
way so that the information can be conveyed from one place to another. There are a number of ways in which a
carrier can be modulated to carry a signal - often an audio signal and the most obvious way is to vary its amplitude.

Amplitude Modulation has been in use since the very earliest days of radio technology. The first recorded instance
of its use was in 1901 when a signal was transmitted by a Canadian engineer named Reginald Fessenden. To achieve
this, he used a continuous spark transmission and placed a carbon microphone in the antenna lead. The sound waves
impacting on the microphone varied its resistance and in turn this varied the intensity of the transmission. Although
very crude, signals were audible over a distance of a few hundred metres. The quality of the audio was not good
particularly as a result of the continuous rasping sound caused by the spark used for the
transmission.

Later, continuous sine wave signals could be generated and the audio quality was greatly improved. As a result,
amplitude modulation, AM became the standard for voice transmissions.

It remains in use today in many forms of communication; for example it is used in portable two way radios, VHF
aircraft radio, Citizen's Band Radio, and in computer modems (in the form of QAM). "AM" is often used to refer to
medium wave AM radio broadcasting.
Amplitude Modulation is defined as a process in which the amplitude of the carrier wave c(t) is varied
linearly with the instantaneous amplitude of the message signal m(t).The standard form of an amplitude modulated
(AM) wave is defined by
Sam(t) = Ac [1 + Ka m(t)] Cos(2пfct)
Sidebands along with the carrier can be represented as follows

Where Ka is constant called the amplitude sensitivity of the modulator. The demodulation circuit is used to
recover the message signal from the incoming AM wave at the receiver. An envelope detector is a simple and yet
highly effective device that is well suited for the demodulation of AM wave, for which the percentage modulation is
less than 100%.Ideally, an envelope detector produces an output signal that follows the envelop of the input signal
wave form exactly; hence, the name. Some version of this circuit is used in almost all commercial AM radio
receivers.

One disadvantage of all amplitude modulation techniques (not only standard AM) is that the receiver amplifies
and detects noise and electromagnetic interference in equal proportion to the signal. Increasing the received signal to
noise ratio, say, by a factor of 10 (a 10 decibel improvement), thus would require increasing the transmitter power
by a factor of 10. This is in contrast to frequency modulation (FM) and digital radio where the effect of such noise
following demodulation is strongly reduced so long as the received signal is well above the threshold for reception.
For this reason AM broadcast is not favored for music and high fidelity broadcasting, but rather for voice
communications and broadcasts (sports, news, talk radio etc.).
To ensure that an amplitude modulated signal does not create spurious emissions outside the
normal bandwidth it is necessary to ensure that the signal does not become over -modulated - this is a conditions that
occurs when the modulation exceeds 100%. At this point the carrier breaks up and intermodulation distortion occurs
leading to large levels of unwanted noise spreading out either side of the carrier and beyond the normal bandwidth.
This can cause interference to other users.

If over-modulation occurs, the carrier becomes phase inverted and this leads to sidebands spreading out either side
of the carrier. The different types of amplitude modulation are
1. double sideband modulation (DSB-AM)
2. double sideband suppressed carrier modulation (DSB SC-AM)
3. single sideband carrier modulation(SSB AM)
4. single sideband suppressed carrier modulation (SSB SC-AM)
5. vestigial sideband modulation (VSB AM)
Applications

Amplitude modulation is used in a variety of applications. Even though it is not as widely used as it was in previous
years in its basic format it can nevertheless still be found.

6. Broadcast transmissions: AM is still widely used for broadcasting on the long, medium and short wave
bands. It is simple to demodulate and this means that radio receivers capable of demodulating amplitude
modulation are cheap and simple to manufacture. Nevertheless many people are moving to high quality
forms of transmission like frequency modulation, FM or digital transmissions.

7. Air band radio: VHF transmissions for many airborne applications still use AM. . It is used for ground to
air radio communications as well as two way radio links for ground staff as well.

8. Single sideband: Amplitude modulation in the form of single sideband is still used for HF radio links.
Using a lower bandwidth and providing more effective use of the transmitted power this form of
modulation is still used for many point to point HF links.
9. Quadrature amplitude modulation: AM is widely used for the transmission of data in everything from
short range wireless links such as Wi-Fi to cellular telecommunications and much more. Effectively it is
formed by having two carriers 90° out of phase.

These form some of the main uses of amplitude modulation. However in its basic form, this form of modulation is
being sued less as a result of its inefficient use of both spectrum and power.

Amplitude modulation advantages & disadvantages

Like any other system of modulation, amplitude modulation has several advantages and disadvantages.
These mean that it is used in particular circumstances where its advantages can be used to good effect.

ADVANTAGES DISADVANTAGES
1. It is simple to implement 1. An amplitude modulation
2. It can be demodulated using a circuit signal is not efficient in
consisting of very few components terms of its power usage

3. AM receivers are very cheap as no 2. It is not efficient in terms of


specialised components are needed. its use of bandwidth,
requiring a bandwidth equal
to twice that of the highest
audio frequency
3. An amplitude modulation
signal is prone to high levels
of noise because most noise is
amplitude based and obviously
AM detectors are sensitive to
it.

In view of its characteristics advantages and disadvantages, amplitude modulation is being used less
frequently. However it is still in widespread use for broadcasting on the long, medium and short wave bands as
well as for a number of mobile or portable communications systems including some aircraft communications.

Power in AM:

Pt=Pc(1+m2/2)
Pt =total power in AM Pc=power in
carrier m=modulation index
Block Diagram

Circuit Diagram:
Demodulator

Procedure:
1. Switch ON the trainer board.
2. Observe the output of RF and AF signal generator using CRO, note that RF voltage is 300mV PP of 1MHz
frequency and AF voltage is 10V PP of 2KHz frequency.

Modulation:
3. Now connect RF and AF signals to the respective inputs of modulation.
4. Initially set both the signals at zero level.
5. Connect one of the input of oscilloscope to modulator output and other input to AF signal.
6. Adjust the RF signal amplitude with the help of potentiometer. So that output of the modulator is 300mV PP by
keeping AF signal at zero level.
7. Now vary the amplitude of AF signal and observe the amplitude modulated wave at output, note the percentage of
modulation for different values of AF signals.

% modulation = ((Vmax - Vmin) / (Vmax + Vmin) )x 100


8. Observe 100% amplitude modulation and over modulation by varying amplitude of the AF signal.

Demodulation:
9. Now connect the modulated output to the demodulator input.
10. Observe demodulated signal at output of demodulator at approximately 50% modulation using oscilloscope.
11. Compare demodulated signal with the original AF signal.
12. And plot the all graphs at 50% modulation

Calculations & Observations

Vmax - Vmin
Modulation Index =
V +V
max min
Tabular Form:
Model Wave Forms
Precautions

1. Avoid loose connections.


2. Avoid parallax error while taking observations.

Result Analysis:
SOFTWARE PROGRAM
SOFTWARE REQUIRED: -
1. MATLAB R2023B
2. Windows 11
Algorithm:
1. Generate message signal
2. Generate carrier signal
3. Apply Mathematical equation of AM.
4. Demodulate AM.
5. Plot all the wave forms.

MATLAB Code:

clc;
clear all;
close all;
Am=1;
Ac=1;
fc=0.4;
fm=0.05;
fs=100;
t=0:(1/fs):200;
ka=1,0.5;
%message signal
msg=Am.*cos(2.*pi.*fm.*t);
%carrier signal
carr=Ac.*cos(2.*pi.*fc.*t);
%AM modulated signal
s=Ac.*(1+ka.*msg).*cos(2.*pi.*fct)
%Demodulated signal
dmsg=amdemod(s,fc,fs);
subplot(4,1,1);
plot(t,msg,'r','linewidth',3);
xlabel('time');ylabel('amplitude';
title('message signal');
subplot(4,1,2);
plot(t,carr,'m','linewidth',3);
xlabel('time');ylabel('amplitude';
title('carrier signal');
subplot(4,1,3);
plot(t,s,'g','linewidth',3);
xlabel('time');ylabel('amplitude';
title('AM signal');
subplot(4,1,4);
plot(t,dmsg,'c','linewidth',3);
xlabel('time');ylabel('amplitude';
title('demodulated signal');
MATLAB output :

HARDWARE GRAPHS:
RESULT:

Hence we observed amplitude modulation and demodulation.


Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment DSB-SC MODULATION AND DEMODULATION


Name of the candidate [Link]
Register Number 2205A41164
Date of Experiment

Experiment-2

DSB – SC MODULATOR AND DETECTOR


Objective
1. To observe the generation and detection processes of the DSB-SC signal using Balanced Modulator and
observe the outputs.
2. To simulate DSB-SC modulation and demodulation
Apparatus Required

□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit


□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
□ CRO with probes □Patch cards □RPS
□Micro phone □Speakers
Theory
Double-sideband suppressed-carrier transmission (DSB-SC) is transmission in which frequencies produced
by amplitude modulation (AM) are symmetrically spaced above and below the carrier frequency and the carrier
level is reduced to the lowest practical level, ideally being completely suppressed.

DSB-SC is basically an amplitude modulation wave without the carrier, therefore reducing power waste, giving it a
50% efficiency. This is an increase compared to normal AM transmission (DSB), which has a maximum efficiency
of 33.333%, since 2/3 of the power is in the carrier which carries no intelligence, and each sideband carries the same
information. Single Side Band (SSB) Suppressed Carrier is 100% efficient.
In the DSB-SC modulation, unlike in AM, the wave carrier is not transmitted; thus, much of the power is
distributed between the sidebands, which implies an increase of the cover in DSB- SC, compared to AM, for the
same power used.

DSB-SC transmission is a special case of double-sideband reduced carrier transmission. It is used for radio data
systems.

Generation: DSB-SC is generated by a mixer. This consists of a message signal multiplied by a carrier signal. The
mathematical representation of this process is shown below, where the product- to-sum trigonometric identity is
used.

Demodulation is done by multiplying the DSB-SC signal with the carrier signal just like the modulation process.
This resultant signal is then passed through a low pass filter to produce a scaled version of original message signal.
DSB-SC can be demodulated by a simple envelope detector, like AM, if the modulation index is less than unity. Full
depth modulation requires carrier re-insertion.
The equation above shows that by multiplying the modulated signal by the carrier signal, the result is a
scaled version of the original message signal plus a second term. Since, this second term is much higher in
frequency than the original message. Once this signal passes through a low pass filter, the higher frequency
component is removed, leaving just the original message.

. With synchronous detection as shown in this applet, the modulated wave (receiving wave) is multiplied with the
carrier frequency which is exactly the same frequency and phase as the transmission carrier wave. For
demodulation, the demodulation oscillator's frequency and phase must be exactly the same as modulation
oscillator's, otherwise, distortion and/or attenuation will occur.

To see this effect, take the following conditions:

1. Message signal to be transmitted:

2. Modulation (carrier) signal:


3. Demodulation signal (with small frequency and phase deviations from the modulation signal):

Two types of generating methods: [Link]


modulator.
1. Ring modulator.
Block Diagram:

Circuit Diagram:
Procedure:
1. Observe the output of the RF generator using CRO. Output should be a sine wave of 100 KHz and
300 mV amplitude.
2. Observe the output of the AF generator using CRO. Output should be a sine wave of 5 KHz and
100mV amplitude.
3. connect the both RF and AF input of the balanced modulator and connect to the output of the RF generator
and AF generator.
1. Connect CRO CH1 input to input of the balanced modulator and CH2 input to the output of the balanced
modulator.
2. Observe the balanced modulator output by slowly increasing the amplitude of AF signal
3. Compare the balanced modulator output with wave form DSB-SC.
4. Connect DSB-SC signal input of the synchronous detector to the output of the balanced modulator and RF input to
the output of the RF generator.

5. Observe the synchronous detector output using CRO and compare with the original AF signal.
OBSERVATIONS:

Model Wave Forms


Precautions

1. Avoid loose connections


2. Avoid parallax error while taking observations.
SOFTWARE PROGRAM:

SOFTWARE REQUIRED: -
1. MATLAB R2023B
2. Windows 11

Algorithm:
1. Generate message signal
2. Generate carrier signal
3. Apply the Mathematical equation of DSB-SC.
4. Demodulate DSB-SC.
5. Plot all the wave forms.
MATLAB Code:
clc;
clear all;
close all;
Am=15;
Ac=3;
fc=0.4;
fm=0.05;
fs=100;
t=0:(1/fs):200;
%message signal
msg=Am.*cos(2.*pi.*fm.*t);
%carrier signal
carr=Ac.*cos(2.*pi.*fc.*t);
%DSB-SC modulated signal
s=msg.*carr;
%Demodulated signal
dmod=amdemod(s,fc,fs);
subplot(4,1,1);
plot(t,msg,'r','linewidth',2);
xlabel('time');ylabel('amplitude');
title('message signal');
subplot(4,1,2);
plot(t,carr,'r','linewidth',2);
xlabel('time');ylabel('amplitude');
title('carrier signal');
subplot(4,1,3);
plot(t,s,'r','linewidth',2);
xlabel('time');ylabel('amplitude');
title('DSB-SC signal');
subplot(4,1,4);
plot(t,dmod,'r','linewidth',3);
xlabel('time');ylabel('amplitude');
title('demodulated signal');

MATLAB Output:
HARDWARE GRAPHS:
RESULT:Hence we observed the DSB-SC modulation and demodulation.
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment FREQUENCY MODULATION AND


DEMODULATION
Name of the candidate [Link]
Register Number 2205A41164
Date of Experiment

Experiment-3
FREQUENCY MODULATION AND DEMODULATION
Objective
1. To observe the process of frequency modulation and demodulation and to calculate the
modulation index (or depth of modulation).
2. To simulate Frequency modulation and demodulation.

Apparatus Required

□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit


□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
□ CRO with probes □Patch cards □RPS
□Micro phone □Speakers

Theory

The process, in which the frequency of the carrier is varied in accordance with the instantaneous amplitude
of the modulating signal, is called “Frequency Modulation”. The FM signal is expressed as

Where Ac is amplitude of the carrier signal, fc is the carrier frequency β is the modulation
Index of the FM wave

Power in the FM/PM signal is relatively easier to calculate; since it is a cosine wave with constant amplitude A, the
average power is given by
The variation in frequency of the FM signal is called the maximum frequency deviation, Δf.

Df fmax - fc (or) Df fc - fmin


= =
Where

fmax = 1 fmin =
1
and
Tmin Tmax
There are two ways to generate a frequency modulation signal, direct FM and indirect FM. With frequency
modulation, the information is incorporated in the frequency, so a nonlinear power amplifier can be used which
ensures good electrical efficiency. Direct FM With direct FM, the modulating signal is input in a VCO (voltage
control oscillator) which generates a frequency shift proportionally to the added voltage. The signal input of the
VCO has a variable capacitance diode, and when a modulating signal voltage is applied here, the capacitance of the
variable capacitance diode changes and the oscillation frequency (carrier frequency) changes
1. The direct methods cannot be used for the broadcast applications. Thus, the alternative method i.e. indirect method
called as the Armstrong method of FM generation is used.

2. In this method the FM is obtained through phase modulation. A crystal oscillator can be used hence the frequency
stability is very high.

Demodulation
The frequency discrimination method or PLL detection method can be used to demodulate the frequency
modulation signal. Demodulation in this applet uses envelope detection after differentiation of the modulated
wave. With frequency modulation, information is carried by the frequency of the carrier wave, so it’s said to be
resistant to noise in the amplitude direction. FM detectors detect frequencies and convert them into voltages, and
since they respond to amplitude fluctuations, these cause errors. The received signal is affected by fading and
outside noise which affects the amplitude direction and appears as distortion or noise in the detector output signal.
Therefore, a limiter circuit is built in before the detector in order to eliminate amplitude fluctuations

There are various reasons why FM is used, but the main reasons are as follows. 1, It has good
tone quality
2, It’s resistant to noise 3,
The hardware is cheap
4, It has a good track record with reliable communication
5, You can set an appropriate modulation factor to improve signal to noise 6, It’s resistant
to fading
7, The technology is well established
8, Component parts are readily available

Applications
Magnetic tape
storage
FM is also used at intermediate frequencies by analog VCR systems (including VHS) to record the
luminance (black and white) portions of the video signal. Commonly, the chrominance component is recorded as a
conventional AM signal, using the higher-frequency FM signal as bias. FM is the only feasible method of recording
the luminance ("black and white") component of video to (and retrieving video from) magnetic tape without
distortion; video signals have a large range of frequency components – from a few hertz to several megahertz, too
wide for equalizers to work with due to electronic noise below −60 dB. FM also keeps the tape at saturation level,
acting as a form of noise reduction; a limiter can mask variations in playback output, and the FM capture effect
removes print-through and pre-echo. A continuous pilot-tone, if added to the signal – as was done on V2000 and
many Hi-band formats – can keep mechanical jitter under control and assist time base correction.

These FM systems are unusual, in that they have a ratio of carrier to maximum modulation frequency of less than
two; contrast this with FM audio broadcasting, where the ratio is around 10,000. Consider, for example, a 6-MHz
carrier modulated at a 3.5-MHz rate; by Bessel analysis, the first sidebands are on 9.5 and 2.5 MHz and the second
sidebands are on 13 MHz and −1 MHz The result is a reversed-phase sideband on +1 MHz; on demodulation, this
results in unwanted output at 6−1 = 5 MHz The system must be designed so that this unwanted output is reduced to
an acceptable level.
Sound:

FM is also used at audio frequencies to synthesize sound. This technique, known as FM synthesis, was
popularized by early digital synthesizers and became a standard feature in several generations of personal computer
sound cards.

Radio:

Edwin Howard Armstrong (1890–1954) was an American electrical engineer who invented wideband
frequency modulation (FM) radio. He patented the regenerative circuit in 1914, the super heterodyne receiver in
1918 and the super-regenerative circuit in 1922. Armstrong presented his paper, "A Method of Reducing
Disturbances in Radio Signaling by a System of Frequency Modulation", (which first described FM radio) before
the New York section of the Institute of Radio Engineers on November 6, 1935. The paper was published in 1936

As the name implies, wideband FM (WFM) requires a wider signal bandwidth than amplitude modulation by an
equivalent modulating signal; this also makes the signal more robust against noise and interference. Frequency
modulation is also more robust against signal-amplitude-fading phenomena. As a result, FM was chosen as the
modulation standard for high frequency, high fidelity radio transmission, hence the term "FM radio" (although for
many years the BBC called it "VHF radio" because commercial FM broadcasting uses part of the VHF band—the
FM broadcast band). FM receivers employ a special detector for FM signals and exhibit a phenomenon known as the
capture effect, in which the tuner "captures" the stronger of two stations on the same frequency while rejecting the
other (compare this with a similar situation on an AM receiver, where both stations can be heard simultaneously).
However, frequency drift or a lack of selectivity may cause one station to be overtaken by another on an adjacent
channel. Frequency drift was a problem in early (or inexpensive) receivers; inadequate selectivity may affect any
tuner.

An FM signal can also be used to carry a stereo signal; this is done with multiplexing and demultiplexing before and
after the FM process. The FM modulation and demodulation process is identical in stereo and monaural processes. A
high-efficiency radio-frequency switching amplifier can be used to transmit FM signals (and other constant-
amplitude signals). For a given signal
strength (measured at the receiver antenna), switching amplifiers use less battery power and typically cost less than a
linear amplifier. This gives FM another advantage over other modulation methods requiring linear amplifiers, such
as AM and QAM.

FM is commonly used at VHF radio frequencies for high-fidelity broadcasts of music and speech. Analog TV sound
is also broadcast using FM. Narrowband FM is used for voice communications in commercial and amateur radio
settings. In broadcast services, where audio fidelity is important, wideband FM is generally used. In two-way radio,
narrowband FM (NBFM) is used to conserve bandwidth for land mobile, marine mobile and other radio services.

There are reports that on October 5, 1924, Professor Mikhail A. Bonch-Bruevich, during a scientific and technical
conversation in the Nizhny Novgorod Radio Laboratory, reported about his new method of telephony, based on a
change in the period of oscillations. Demonstration of frequency modulation was carried out on the laboratory
model.

Block Diagram:
Circuit diagrams:
Procedure:
1. Connections are made as per the circuit diagram.
2. Observe the output of AF generator using CRO. Note that AF voltage is approximately 20V PP of
500Hz frequency.

Modulator:
3. Observe the carrier signal at the modulator output. The carrier signal is approximately 5V PP of
100 KHz frequency.
4. Connect AF signal to the AF input of the modulator.
5. Observe the frequency modulated wave at the modulator output by varying deviation
potentiometer, which varies the amplitude of the incoming AF signal.
6. Set deviation potentiometer in minimum position and calculated the output signal frequency of
modulator which indicates RF frequency i.e fc.
7. Now set the deviation potentiometer to middle position and observe the output signal. Calculate
the maximum frequency i.e, S = fb-fc.
8. Now find the modulation index M – S/Fm.
Fm = Maximum frequency deviated / Modulating frequency.
9. Now set the AF signal to 5 KHz frequency.
10. Connect one of the input of oscilloscope to the modulator output and another input to AF generator.
11. Observe trace, the frequency modulated wave and AF signal simultaneously.

Demodulator:
12. Connect the modulator output to the demodulator input.
13. Observe trace the demodulated signal at the output of modulator and compare it with the original
AF signal.

Calculations

Modulation Index Df
= f
m
where fm = Modulating signal frequency
Δf = Frequency deviation
D fmax - fc = fc - fmin
(or) Df
f
=
Where

fmax = 1 fmin =
1
and
Tmin Tmax

Modulating Signal Generator:

Amplitude =4*5v=20v Time Period =210milli sec Frequency =4.7KHZ

Carrier Signal Generator:

Amplitude =4.6v Time Period =11milli sec Frequency =90KHZ

Demodulated Output:

Amplitude =20v Time Period =210milli sec Frequency =4.7KHZ


Observations
Waveforms:
Precautions
1. Avoid loose connections.
2. Avoid parallax error while taking observations.

Result Analysis:
SOFTWARE PROGRAM:
SOFTWARE REQUIRED: -
1. MATLAB R2023B
2. Windows 11

Algorithm:
1. Generate message signal
2. Generate carrier signal
3. Apply Mathematical equation of FM.
4. Demodulate FM.
5. Plot all the wave forms.

MATLAB Code:
clc;
clear all;
close all;
%Am=5;
%Ac=2;
fc=10;
fm=1;
fs=70;
fdev=5;
t=0:0.01:3;
%message signal
msg=cos(2.*pi.*fm.*t);
%carrier signal
carr=cos(2.*pi.*fc.*t);
%FM modulated signal
s=fmmod(msg,fc,fs,fdev);
%FM Demodulated signal
dmod=fmdemod(s,fc,fs,fdev);
subplot(4,1,1);
plot(t,msg,'r','linewidth',3);
xlabel('time');ylabel('amplitude');
title('message signal');
subplot(4,1,2);
plot(t,carr,'g','linewidth',3);
xlabel('time');ylabel('amplitude');
title('carrier signal');
subplot(4,1,3);
plot(t,s,'y','linewidth',3);
xlabel('time');ylabel('amplitude');
title('FM signal');
subplot(4,1,4);
plot(t,dmod,'k','linewidth',3);
xlabel('time');ylabel('amplitude');
title('demodulated signal');

MATLAB Output:
HARDWARE GRAPHS:
Result Analysis:

Hence we observed frequency modulation and demodulation.


Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment Pulse amplitude modulation and demodulation


Name of the candidate [Link]
Register Number 2205a41164
Date of Experiment

Experiment-4
PULSE AMPLITUDE MODULATION & DEMODULATION

Objective:
1. To study the operation of Pulse Amplitude modulation & Demodulation and plot the waveforms.
2. To simulate Pulse amplitude modulation and demodulation

Apparatus Required

1. AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit


□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
2. CRO with probes □Patch cards □RPS
□Micro phone □Speakers
Theory:
Pulse amplitude modulation (PAM) is the transmission of data by varying the amplitudes (voltage or power
levels) of the individual pulses in a regularly timed sequence of electrical or electromagnetic pulses. The number of
possible pulse amplitudes can be infinite (in the case of analog PAM), but it is usually some power of two so that the
resulting output signal can be digital
. For example, in 4-level PAM there are 4 possible discrete pulse amplitudes; in 8-level PAM there are 2^3 possible
discrete pulse amplitudes; and in 16-level PAM there are 2^4 possible discrete pulse amplitudes.

In some PAM systems, the amplitude of each pulse is directly proportional to the instantaneous modulating-signal
amplitude at the time the pulse occurs. In other PAM systems, the amplitude of each pulse is inversely proportional
to the instantaneous modulating-signal amplitude at the time the pulse occurs. In still other systems, the intensity of
each pulse depends on some characteristic of the modulating signal other than its strength, such as its instantaneous
frequency or phase .
There are two types of pulse amplitude modulation:

1. Single polarity PAM: In this a suitable fixed DC bias is added to the signal to ensure that all the pulses are
positive.
2. Double polarity PAM: In this the pulses are both positive and negative.

Pulse-amplitude modulation is widely used in modulating signal transmission of digital data, with non-baseband
applications having been largely replaced by pulse-code modulation, and, more recently, by pulse-position
modulation.

In particular, all telephone modems faster than 300 bit/s use quadrature amplitude modulation (QAM). (QAM uses a
two-dimensional constellation).

The number of possible pulse amplitudes in analog PAM is theoretically infinite. Digital PAM reduces the number
of pulse amplitudes to some power of two. For example, in 4-level PAM there

are possible discrete pulse amplitudes; in 8-level PAM there are possible discrete pulse amplitudes; and in 16-level
PAM there are possible discrete pulse amplitudes.

In PAM the amplitude of the message or modulating signal is mapped to a series of pulses with two possible variant
1. Flat Top PAM:- The amplitude of each pulse is directly proportional to instantaneous modulating signal amplitude
at the time of pulse occurrence and then keeps the amplitude of the pulse for the rest of the half cycle.
2. 2) Natural PAM:- The amplitude of each pulse is directly proportional to the instantaneous modulating signal
amplitude at the time of pulse occurrence and then follows the amplitude of the modulating signal for the rest of
the half cycle.

Advantages of PAM:

PAM allows data to be transmitted more effectively, efficiently and quickly using
conventional copper wires in greater volume.
The frequency modulations available are infinite; hence PAM formulas can be developed
continually to allow increased data throughput over existing networks.
PAM is also the simplest form of modulation.
Applications:
The PAM modulation technique is widely used in high speed digital communications like telephone modems. They
are used to drive LED lights more efficiently than using PWM method. Unlike the PPM the transmitter and receiver
synchronization is not required for the PAM.

They are also used in Ethernet. For example, 100BASE-T2 – operating at 100Mb/s – Ethernet medium is using 5
level PAM modulations running at 25 mega pulses/sec over four wires. Later developments include the 100BASE-T
medium which raised the bar to 4 wire pairs, running each at 125 mega pulses/sec in order to achieve 1000 Mbps
data transfer rates, but still with the same PAM5 for each pair.

More recently, PAM12 and PAM8 have gained consideration in the newly proposed IEEE 802.3an standard for
10GBase-T — ten gigabyte Ethernet over copper wire.

Block Diagram:
Circuit Diagrams:
PAM Modulator Circuit Diagram:

PAM De-modulator Circuit Diagram:

Experimental Procedure:

1. As the circuitry is already wired you just have to trace the circuit according to the circuit diagram given
above.
2. Connect trainer to mains and switch on the power.
3. Observe the output of AF generator and pulse generator using CRO and note that AF signal is
approximately 3V P-P of 400Hz frequency and pulse generator output is pulse train of 10V P-P with
frequency between 1 KHz and 6 KHz.
Modulator:
4. Connect pulse output and AF output to the respective inputs of modulator circuit.
5. Connect one of the inputs of oscilloscope to the modulator output and another to AF signal.
6. Initially set the amplitude of the AF generator to minimum level and sampling frequency to 1 KHz (by
adjusting the pre-set provided in pulse generator block). Note down the output of modulator, by varying
amplitude of modulating signal observe the modulator output so that you can notice the amplitude of the
sampling pulses is varying in accordance with the modulating signal.
Demodulator:
7. Connect PAM wave input to demodulator input and set sampling pulse frequency to maximum (6 KHz).
8. Observe demodulated signals at output of demodulator; compare it with the original AF signal.
(Note: Only shape, amplitude will be attenuated).
9. You can observe the amplified signal by applying demodulated signal to amplifier.
10. Find the detected signal is same as the AF signal applied. Thus no information is lost in the process of
modulation.

CALCULATIONS
Waveforms:
Precautions
1. Avoid loose connections.
2. Avoid parallax error while taking observations.

SOFTWARE PROGRAM:
SOFTWARE REQUIRED: -
1. MATLAB R2023B
2. Windows 11

Algorithm:
1. Generate a message signal with a suitable frequency.
[Link] a suitable sampling frequency and sample the above signal.
[Link] the PAM waveform.

MATLAB Code:
clc
clear all

close all

fm=10

fc=100

fs=100*fm

t=0:1/fs:0.4

msg=cos(2*pi*fm*t)

car=0.5*square(2*pi*fc*t)+0.5

s=msg.*car

tt=[];

%single sided PAM signal

for i=1:length(s);

if s(i)==0

tt=[tt,s(i)];

else

tt=[tt,s(i)+2];

end

end

%Demodulation

[b a]=butter(2,0.02)

y=filter(b,a,s)

subplot(5,1,1)

plot(t,msg)

xlabel('t')

ylabel('amp')

title('msg signal')

subplot(5,1,2)

plot(t,car)
xlabel('t')

ylabel('amp')

title('carrier pulse')

subplot(5,1,3)

plot(t,s)

xlabel('t')

ylabel('amp')

title('PAM signal')

subplot(5,1,4)

plot(t,tt)

xlabel('t')

ylabel('amp')

title('PAM signal2')

subplot(5,1,5)

plot(t,y)

xlabel('t')

ylabel('amp')

title('demodulated signal')
MATLAB Output:

HARDWARE GRAPHS:
RESULT
Hence we observed pulse amplitude modulation and demodulation.
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment TIME DIVISION MULTIPLEXING AND


DEMULTIPLEXING
Name of the candidate [Link]
Register Number 2205A41164
Date of Experiment
Experiment-5
TIME DIVISION MULTIPLEXING & DEMULTIPLEXING
Objective
• To study Time Division Multiplexing and De-multiplexing, using Pulse Amplitude Modulation and
Demodulation and to reconstruct the signals at the Receiver, using Filters.
• To simulate Time-division multiplexing (TDM) and de-multiplexing.

Apparatus Required

□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit


□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
□ CRO with probes □Patch cards □RPS
□Micro phone □Speakers
Theory:

Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a
common signal path by means of synchronized switches at each end of the transmission line so that each signal
appears on the line only a fraction of time in an alternating pattern. It is used when the data rate of the transmission
medium exceeds that of signal to be transmitted. This form of signal multiplexing was developed in
telecommunications for telegraphy systems in the late 19th century, but found its most common application in
digital telephony in the second half of the 20th century

Time-division multiplexing is used primarily for digital signals, but may be applied in analog multiplexing in
which two or more signals or bit streams are transferred appearing simultaneously as sub-channels in one
communication channel, but are physically taking turns on the channel. The time domain is divided into several
recurrent time slots of fixed length, one for each sub-channel.
A sample byte or data block of sub-channel 1 is transmitted during time slot 1, sub-channel 2

during time slot 2, etc. One TDM frame consists of one time slot per sub-channel plus a synchronization channel
and sometimes error correction channel before the synchronization. After the last sub-channel, error correction, and
synchronization, the cycle starts all over again with a new frame, starting with the second sample, byte or data block
from sub-channel 1, etc

TDM can be further extended into the time-division multiple access (TDMA) scheme, where several stations
connected to the same physical medium, for example sharing the same frequency channel, can communicate.
Application examples include:

1. The GSM telephone system


2. The Tactical Data Links Link 16 and Link 22

Types of TDM :

1. Synchronous TDM

2. Asynchronous TDM

In synchronous TDM, each device is given same time slot to transmit the data over the link, irrespective of the fact
that the device has any data to transmit or not. Hence the name Synchronous TDM. Synchronous TDM requires that
the total speed of various input lines should not exceed the
capacity of path

Asynchronous TDM

1. It is also known as statistical time division multiplexing.

2. Asynchronous TDM is called so because is this type of multiplexing, time slots are not fixed
i.e. the slots are flexible.

3. Here, the total speed of input lines can be greater than the capacity of the path.

4. In synchronous TDM, if we have n input lines then there are n slots in one frame. But in asynchronous it is not
so.

Advantages of TDM :

1. Full available channel bandwidth can be utilized for each channel.


2. Inter modulation distortion is absent.
3. TDM circuitry is not very complex.
4. The problem of crosstalk is not severe.

Disadvantages of TDM :

1. Synchronization is essential for proper operation.


2. Due to slow narrowband fading, all the TDM channels may get wiped out.

Applications:

The T1 system for wireline telephone networks


The T1 system is used for wireline long-distance service in North America and is an excellent example of
TDM. Speech from a telephone conversation is sampled once every 125 msec and each sample is converted into
eight bits of digital data (see Chapter 8 for more details). Using this technique, a transmission speed of 64,000
bits/sec is required to transmit the speech. A T1 line is essentially a channel capable of transmitting at a speed of
1.544 Mbit/sec. This is a much higher
transmission speed than a single telephone conversation needs, so TDM is used to allow a single T1 line to
carry 24 different speech signals between, say, two different telephone substations (called central offices) within a
city. As shown in Figure . the 1.544 Mbit/sec bit stream is divided into 193-bit frames. The duration of each frame is

corresponding to the period between samples of the speech. Each frame is divided into 24 slots, which are each eight
bits wide (corresponding to the number of bits needed to digitize a speech sample). One additional bit at the end of
the frame is used for signaling. The eight bits of data corresponding to a sample of the speech are placed into one of
the 24 slots in the frame.

For longer distances (say, between two large cities) higher-capacity channels are used and multiple T1 lines are time
division multiplexed onto the new channels. A T3 channel for example, has a transmission speed of 44.736 Mbit/sec
and uses TDM to carry 28 T1 lines (a total of 672 different speech signals) plus signalling. For more information on
this hierarchical multiplexing system.

Fig: Time division multiplexing on a T1 line.

TDM with sources having different data rates


Consider the case of three streams with bit rates of 8 kbit/sec,16 kbit/sec, and 24 kbit/sec, respectively. We want
to combine these streams into a single high-speed stream using TDM. The high-speed stream in this case must have
a transmission rate of 48 kbit/sec, which is the sum of the bit rates of the three sources. To determine the number of
time slots to be assigned to each
source in the multiplexing process. we must reduce the ratio of the rates, 8:16:24, to the lowest possible form,
which in this case is 1:2:3. The sum of the reduced ratio is 6, which will then represent the minimum length of the
repetitive cycle of slot assignments in the multiplexing process. The solution is now readily obtained: In each cycle
of six time slots we assign one slot to Source A (8 kbit/sec), two slots to Source B (16 kbit/sec), and three slots to
Source: C (24 kbit/sec). Figure 7-4 illustrates this assignment, using “a” to indicate data from Source A, “b” to
indicate data from Source B, and “c” to indicate data from Source C.

Figure—Multiplexing input lines with different transmission speeds.

A more complex TDM system

1. Consider a system with four low-bit-rate sources of 10 kbit/sec, 15 kbit/sec, 20 kbit/sec, and 30 kbit/sec. Determine
the slot assignments when the data streams are combined using TDM.

Solution

2. The rate ratio 10:15:20:30 reduces to 2:3:4:6. The length of the cycle is therefore 2 + 3 + 4
+ 6 = 15 slots. Within each cycle of 15 slots, we assign two slots to the 10 kbit/sec source, three slots to the
15 kbit/sec source, four slots to the 20 kbit/sec source, and six slots to the 30 kbit/sec source.

3. So far we have considered a form of TDM that is based on fixed slot assignments to each of the low-bit-rate data
streams. In other words, each stream has predefined slot positions in the combined stream, and the receiver must be
aware which slots belong to which input stream. Both transmission ends, the transmitter and the receiver, must be
perfectly synchronized to the slot period. For this reason, the technique is usually called synchronous TDM.

4. There is another important version of TDM, usually referred to as statistical TDM. Statistical TDM is useful for
applications in which the low-bit-rate streams have speeds
that vary in time. For example, a low-bit-rate stream to a single terminal in a computer network may fluctuate
between 2 kbit/sec and 50 kbit/sec during an active connection session (we've all seen variable speeds
during Internet connections, for instance). If we assign the stream enough slots for its peak rate (that is, for
50 kbit/sec), then we will be wasting slots when the rate drops well below the peak value. This waste can
be especially significant if the system has many variable-speed low-bit-rate streams.

5. Statistical TDM works by calculating the average transmission rates of the streams to be combined, and then uses a
high-speed multiplexing link with a transmission rate that is equal to (or slightly greater than) the statistical average
of the combined streams. Since the transmission rates from each source are variable, we no longer assign a fixed
number of time slots to each data stream. Rather, we dynamically assign the appropriate number of slots to
accommodate the current transmission rates from each stream. Because the combined rate of all the streams will
also fluctuate in time between two extreme values, we need to buffer the output of the low-bit-rate streams when
the combined rate exceeds the transmission rate of the high-speed link.

6. With statistical TDM, we are no longer relying on synchronized time slots with fixed assignments for each input
stream, as we did with synchronous TDM. So how does the demultiplexer in statistical TDM know which of the
received bits belongs to which data stream? Prior to transmission, we divide each stream of bits coming from a
source into fixed-size blocks. We then add a small group of bits called a header to each block, with the header
containing the addresses of the source and intended user for that block. The block and the header are then
transmitted together across the channel. Combined, the block and header are called a packet.

7. Actually, the header may contain other information besides the source and user addresses, such as extra bits for
error control or additional bits for link control (used, for example, to indicate the position of a particular block in a
sequence of blocks coming from the same user, or to indicate priority level for a particular message). Extra bits can
also be added to the beginning and end of a block for synchronization; a particular pattern of bits, called a start
flag, can be used in the header to mark the start of a block, and another particular pattern of bits, called an end flag,
can be used to conclude the block. Each block transmitted
across the channel thus contains a group of information bits that the user wants, plus additional bits needed by the
system to ensure proper transmission. These additional bits, while necessary to system operation, reduce
the effective transmission rate on the channel. Figures present the statistical TDM technique and the
structure of a typical packet.

1. Figure 7-5—Statistical TDM.

·
Block Diagram:

Procedure for Modulation:


2. Connect the four channel input 250 KHz, 500 KHz, 1 KHz, 2 KHz to the input of transmitter
CH0, CH1, CH2 and CH3 respectively.
3. Connect the TX clock transmitter clock to RX clock (Receiver clock).
4. Connect the TX CH0 (Transmitter Sync) to RX CH0 (Receiver Sync)
5. Connect the TXD (Transmitter Data) to RXD(receiver Data).
6. Observe the multiplexed Data at TDX

Procedure for Demodulation:

7. Connect the TDX output the RDX Receiver.


8. Observe the Demultiplexed signals at the receiver across the output of fourth order low pass
filter at CH0, CH0, CH1, CH2, and CH3 respectively.
Calculations:
Transmitter side:
Message signal (CH0)
Amplitude =4V Time Period =4msec Frequency =250HZ

Message signal (CH1)


Amplitude =4v Time Period =2msec Frequency =500HZ

Message signal (CH2)


Amplitude =4v Time Period =1msec Frequency =1000HZ

Message signal (CH3)


Amplitude =4v Time Period=0.5msec Frequency =2000HZ

Receiver side:
De-multiplexed signal (OUT0)
Amplitude =4v Time Period =4msec Frequency =250HZ

De-multiplexed signal (OUT1)


Amplitude =4v Time Period =2msec Frequency =500HZ

De-multiplexed signal (OUT2)


Amplitude =4v Time Period =1msec Frequency =1000HZ

De-multiplexed signal (OUT3)


Amplitude =4v Time Period =0.5msec Frequency =2000HZ
Expected Graph:
Wave forms:
Precautions
1. Avoid loose connections.
2. Avoid parallax error while taking observations.

Result Analysis:
SOFTWARE PROGRAM

SOFTWARE REQUIRED: -
1. MATLAB R2023B
2. Windows 11

Algorithm:
[Link] sinusoidal signal with suitable frequency
[Link] triangular signal with suitable frequency [Link]
sampling to both signals.
[Link] both the signals.
[Link] both the signals.
1. plot all the waveforms.

MATLAB Code:
clc;
clear all;
close all;
x=0:0.5:4*pi;
sig1=8.*sin(x);
l1=length(sig1);
sig2=8.*triang(l1);
subplot(2,2,1);
plot(sig1);
xlabel('time');
ylabel('amplitude');
title('sig1 in continuous form');
grid on;
subplot(2,2,2);
plot(sig2);
xlabel('time');
ylabel('amplitude');
title('sig2 in continuous form');
subplot(2,2,3);
stem(sig1);
xlabel('time');
ylabel('amplitude');
title('sig1 in discrete form');
subplot(2,2,4);
stem(sig2);
xlabel('time');
ylabel('amplitude');
title('sig2 in discrete form');
l1=length(sig1);
l2=length(sig2);
%muliplexing
for i=1:l1
sig(1,i)=sig1(i);
sig(2,i)=sig2(i);
end;
tdmsig=reshape(sig,1,2*l1);
figure;
stem(tdmsig);
xlabel('time');
ylabel('amplitude');
title('TDM signal');
%demultiplexing
demux=reshape(tdmsig,2,l1);
l2=length(sig2);
for i=1:l1
sig3(i)=demux(1,i);
sig4(i)=demux(2,i);
end
figure;
plot(sig3);
xlabel('time');
ylabel('amplitude');
title('recovered sine signal');
figure;
plot(sig4);
xlabel('time');
ylabel('amplitude');
title('recovered triangular signal');

MATLAB Output:
HARDWARE GRAPHS:
RESULT:
Hence we observed time division multiplexing and demultiplexing.
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment Generation and detection of FSK


Name of the candidate [Link]
Register Number 2205a41164
Date of Experiment
Experiment-6
FREQUENCY SHIFT KEYING: GENERATION AND DETECTION
1. To study the operation of FSK (binary) modulation & Demodulation and to plot the FSK waveforms for
Binary data at different frequencies.
2. To Simulate FSK using MATLAB.

EQUIPMENT REQUIRED
• Frequency shift keying system trainer.
• Dual trace oscilloscope.
• Digital multimeter.
• Digital frequency counter.

THEORY
Frequency-shift keying (FSK) is a frequency modulation scheme in which digital information is
transmitted through discrete frequency changes of a carrier signal. The technology is used for communication
systems such as amateur radio, caller ID and emergency broadcasts. The simplest FSK is binary FSK (BFSK).
BFSK uses a pair of discrete frequencies to transmit binary (0s and 1s) information. With this scheme, the "1" is
called the mark frequency and the "0" is called the space frequency. The time domain of an FSK modulated carrier
is illustrated in the figures to the right. Frequency Shift Keying (FSK) modulation is a popular form of digital
modulation used in low-cost applications for transmitting data at a moderate or low rate over wired as well as
wireless channels. In general, an M-ary FSK modulation scheme is a power efficient modulation scheme and several
forms of M-ary FSK modulation are becoming popular for spread spectrum communications and other wireless
applications. In this lesson, our discussion will be limited to binary frequency shift keying (BFSK). Two carrier
frequencies are used for binary frequency shift keying modulation. One frequency is called the ‘mark’ frequency (f2)
and the other as the space frequency (f 1). By convention, the ‘mark’ frequency indicates the higher of the two
carriers used. If Tb indicates the duration of one information bit.
Fig.5.1. FSK modulator

Fig.5.2. FSK demodulator


Advantages
1. It promotes long-distance communications.
2. High security.
3. Low noise.
4. Efficiency is high.
5. Decoding of the signal is easy.
Disadvantage
It requires high bandwidth.
Applications
1. Point to point military communications.
2. Telephone Modem.
BLOCK DIAGRAM

Fig.5.3. FSK modulator using XR 2206

Fig.5.4. FSK De-modulator using LM 565

PROCEDURE
1. Study the theory of operation.
2. Connect the trainer to mains and switch on the power supply.
3. Observe the data signals using CRO. It should be a square wave of 20Hz to 200 Hz @ 10Vp-p.
4. Connect the output of the logic source to the data input of the FSK modulator.
5. Set logic source switch in 0 positions.
6. Connect FSK modulator output to CRO as well as a frequency counter (Multimeter in Hz mode).
7. Set the output frequency of the FSK modulator as per your desire (say 1.2KHz) with the help of control F0 which
represents logic 0.
8. Set logic source switch in 1 position.
9. Set the output frequency of the FSK modulator as per your desire (say 2.4 KHz) with the help of control F1 which
represents logic 1.
Note: we have chosen F0 as 1.2khz and F1 as 2.4 KHz for each of operation, in fact, you may set any value.
10. Now connect the output of the data signal generator to data input of the FSK modulator.
11. Keep CRO in dual mode connect CH1 input of the oscilloscope to the input of the FSK modulator and CH2 input
to the output of the FSK modulator.
12. Observe the FSK signal for different data signal frequencies and plot them. By this, we can observe that the carrier
frequency is shifting between two signals.
13. Now connect the output of FSK modulator to the input of FSK demodulator and the observe the output FSK
demodulator, it should be same as data signal.
WAVEFORMS

Fig.5.5. Waveforms of FSK

CALCULATIONS
OBSERVATION TABLE
Table 5.1. The tabular column to note down the Amplitude, frequency and time period of FSK waveform values.
S. No. Type of signal Amplitude (Volts) Frequency (Hz) Time period (Sec)

1 Message signal 2x2=4V 200Hz 5msec

2 Carrier signal 1 2.2x2=4.4v 714.2Hz 1.4msec


3 Carrier signal 2 2.2x2=4.4v 2777Hz 0.36msec

4 Modulating 2.2x2=4.4v - -
signal
5 Demodulated 2x2=4v 200Hz 5msec
signal

RESULT
Precautions
14. Avoid loose connections.
15. Avoid parallax error while taking observations.

Result Analysis:
SOFTWARE PROGRAM

SOFTWARE REQUIRED: -
1. MATLAB R2023b
2. Windows 11

MATLAB Code:
clc;
clear all;
close all;
%message signal
a=1;
fm=10;
t=0:0.001:1;
msg=(a.*square(2.*pi.*fm.*t,50)+a)./2;
%carrier1 signal
fc1=20;
carr1=cos(2.*pi.*fc1.*t);
%carrier2 signal
fc2=30;
carr2=cos(2.*pi.*fc2.*t);
%fsk signal
for i=0:1000
if msg(i+1)==1
fsk(i+1)=carr1(i+1);
else
fsk(i+1)=carr2(i+1);
end;
end;
%Demodulation
for j=1:length(t)
if(carr1(j)==fsk(j))
demod(j)=1;
else
demod(j)=0;
end;
end;
subplot(5,1,1)
plot(t,msg,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('message signal');
subplot(5,1,2)
plot(t,carr1,'c','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('carrier1 signal');
subplot(5,1,3)
plot(t,carr2,'k','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('carrier2 signal');
subplot(5,1,4)
plot(t,fsk,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('FSK signal');
subplot(5,1,5)
plot(t,demod,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('demodulated signal');
Output

HARDWARE GRAPHS:
RESULT:
Hence we observed Frequency Shift Keying generation and detection..
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment PSK generation and detection


Name of the candidate [Link]
Register Number 2205a41164
Date of Experiment
Experiment-7
PHASE SHIFT KEYING: GENERATION AND DETECTION
OBJECTIVE
1. To study the operation of PSK modulation & demodulation and to plot the PSK waveforms for binary data at
different frequencies.
2. To Simulate PSK using MATLAB.
EQUIPMENT REQUIRED
1. Phase shift keying system trainer.
2. Dual trace oscilloscope.
3. Digital multimeter.

THEORY
Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing (modulating) the
phase of a reference signal (the carrier wave). The modulation occurs by varying the sine and cosine inputs at a
precise time. It is widely used for wireless LANs, RFID, and Bluetooth communication. Any digital modulation
scheme uses a finite number of distinct signals to represent digital data. PSK uses a finite number of phases, each
assigned a unique pattern of binary digits. Usually, each phase encodes an equal number of bits. Each pattern of bits
forms the symbol that is represented by the particular phase. The demodulator, which is designed specifically for the
symbol-set used by the modulator, determines the phase of the received signal and maps it back to the symbol it
represents, thus recovering the original data. This requires the receiver to be able to compare the phase of the
received signal to a reference signal — such a system is termed coherent (and referred to as CPSK).

In this kind of modulation, the sine carrier takes 2 or more phase values, directly determined by the binary
data signal (2-phase modulation) or by the combination of a certain number of bits of the same data signal (N-phase
modulation). In 2-phase PSK modulation, called 2-PSK, or Binary PSK (BPSK), or Phase Reversal Keying (PRK),
the sine carrier takes 2 phase values, determined by the binary data signal. A modulation technique is the one using a
balanced modulator. The output sine-wave of the modulator is the direct or inverted (i.e. shifted of 180°) input
carrier, a function of the data signal. Constellation Diagram The modulation states of the PSK Modulator are
represented with points in a vectoral diagram. Each point is a modulation state, characterized by a phase and an
amplitude. This representation is called a constellation diagram, or more simply constellation.

Fig.6.1. PSK Modulation


Fig.6.2. PSK Demodulation
Main aspects
The main aspects characterizing the 2-PSK are:
1. use of digital radio transmission
2. it requires circuits of average-high complexity
3. high possibility of error but lower than the FSK
4. if Fb is the bit transmission speed, the minimum spectrum Bw of the modulated signal is higher than Fb
5. the transmission efficiency, defined as the ratio Fb and Bw, is lower than 1
6. the Baud or Baud rate, defined as the Modulation speed or symbol speed, is equal to the transmission speed Fb.
Advantages
1. The probability of error is very less.
2. Low noise
3. High data rates used in long distance communications.
4. Good sensitivity.
Disadvantage
PSK has more complex signal detection/recovery process than in ASK and FSK.
Applications
It is widely used for wireless LANs, RFID, and Bluetooth communication.

CIRCUIT DIAGRAM
PSK Modulator:

Fig.6.3. PSK Modulator


PSK De-Modulator:
Fig.6.4. PSK De-Modulator

PROCEDURE
1. Study the theory of operation.
2. Connect the trainer to mains and switch on the power supply.
3. Observe the output of the carrier generator using CRO; it should be an 8KHz sine with 5Vpp amplitude.

Modulation:
4. Observe the various data signals (1KHz, 2KHz, and 4KHz) using CRO.
5. Connect carrier signal to the carrier input of the PSK modulator.
6. Connect data signal say 4KHz from the data source to the data input of the modulator.
7. Keep CRO in dual mode.
8. Connect the CH1 input of the CRO to the data signal and CH2 to the output of the PSK modulator.
9. Observe the PSK o/p signal with respect to the data signal and plot the waveforms. compare the plotted waveforms
with the given waveforms.

De-Modulation:
10. Connect the PSK input of the demodulator.
11. Connect the carrier to the carrier input of the PSK demodulator.

Note: In the actual communication system reference carrier is generated at the receiver.
12. Keep CRO in dual mode.
13. Connect CH1 to the data signal (at the modulator) and CH2 to the output of the demodulator.
14. Compare the demodulator signal with the original data signal. By this, we can notice that there is no loss in the
modulation and demodulation process.
15. Repeat steps 7 to 15 with different data signals i.e.2KHz and 1KHz.

WAVEFORMS

Fig.6.5. Waveforms of PSK

CALCULATIONS
OBSERVATION TABLE
Table 6.1. The tabular column to note down the Amplitude, frequency and time period of PSK waveform values.
SOFTWARE PROGRAM:
Software Required:
MATLAB R2023b

MATLAB Code:
clc;
clear all;
close all;
%message signal
a=1;
fm=10;
t=0:0.001:1;
fc=20;
msg=(a.*square(2.*pi.*fm.*t,50)+a)/2;
%carrier signal
carr=sin(2.*pi.*fc.*t);
ask=msg.*carr
for j=1:length(t)
if(carr(j)==ask(j))
demod(j)=1;
else
demod(j)=0;
end;
end;
subplot(4,1,1)
plot(t,msg,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('message signal');
subplot(4,1,2)
plot(t,carr,'c','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('carrier signal');
subplot(4,1,3)
plot(t,ask,'k','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('ASK signal');
subplot(4,1,4)
plot(t,demod,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('demodulated signal');

output;

HARDWARE GRAPHS:
RESULT:
Hence we observed phase shift keying modulation and demodulation.
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment ASK generation and detection


Name of the candidate [Link]
Register Number 2205a41164
Date of Experiment
Experiment-8
AMPLITUDE SHIFT KEYING: GENERATION AND DETECTION

[Link] perform the techniques of ASK Generation and Detection.


[Link] simulate ASK using MATLAB.

EQUIPMENT REQUIRED
 ASK Trainer kit.
 Dual trace oscilloscope
 Power supply
 Patch chords.

Fig.8.1. ASK Modulator


Fig. 8.2. ASK Modulator and Demodulator

PROCEDURE:
Modulation:

• Study the theory of ASK operation.


• Connect the trainer to mains and switch on the power supply.
• Observe the output of the carrier generator using CRO; it should be 10KHz sine with 5Vpp amplitude.
• Observe the data signals using CRO, it should be 200Hz square wave with 5Vpp amplitude.
• Connect carrier signal to the carrier input of the ASK modulator.
• Connect data signal say from the modulating signal source to the data input of the modulator.
• Keep CRO in dual mode.
• Connect the CH1 input of the CRO to the data signal and CH2 to the output of the ASK modulator.
• Observe the ASK o/p signal with respect to the data signal and plot the waveforms. compare the plotted
waveforms with the given waveforms.

De-Modulation:
• Connect the ASK input to the demodulator.
• Keep CRO in dual mode.
• Connect CH1 to the data signal (at modulator) and CH2 to the output of the demodulator.
• Compare the demodulator signal with the original data signal.

Waveforms:

Fig. 8.3. ASK Waveforms


SOFTWARE PROGRAM:
Software Required:
MATLAB R2023b

MATLAB Code:

clc;
clear all;
close all;
%message signal
a=1;
fm=5;
t=0:0.001:1;
fc=50;
msg=(a.*square(2.*pi.*fm.*t,50)+a)/2;
%carrier signal
carr=sin(2.*pi.*fc.*t);
ask=msg.*carr
for j=1:length(t)
if(carr(j)==ask(j))
demod(j)=1;
else
demod(j)=0;
end;
end;
subplot(4,1,1)
plot(t,msg,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('message signal');
subplot(4,1,2)
plot(t,carr,'c','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('carrier signal');
subplot(4,1,3)
plot(t,ask,'k','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('ASK signal');
subplot(4,1,4)
plot(t,demod,'r','linewidth',3);
xlabel('time')
ylabel('amplitude');
title('demodulated signal');

OUTPUT:

HARDWARE GRAPHS:
RESULT:
Hence we observed amplitude shift keying generation and detection.
Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment Verification of sampling theorem


Name of the [Link]
candidate
Register Number 2205a41164
Date of Experiment
Experiment-9
SAMPLING THEOREM

3. To perform the sampling theorem verification.


4. To simulate the sampling theorem verification.

EQUIPMENT REQUIRED

5. Sampling Theorem/PAM-TDM Trainer kit.


6. Dual trace oscilloscope
7. Power supply
8. Patch chords.

BLOCK DIAGRAM: (Use screen shot of the block diagram from observation book)
RESULT
SOFTWARE PROGRAM:

Software Required:
MATLAB R2023b

clc;
clear all;
close all;
t=-5:0.001:5;
f1=3;f2=25;
x=cos(2.*pi.*f1.*t)+cos(2.*pi.*f2.*t)
figure(1);
plot(t,x);
axis([-0.4 0.4 -2 2])
xlabel('time');
ylabel('amplitude');
title('message signal');
grid on;
%under sampling
fs1=0.5*f2;
ts1=1/fs1;
n1=-0.4:ts1:0.4;
xs1=cos(2.*pi.*f1.*n1)+cos(2.*pi.*f2.*n1);
figure(2);
stem(n1,xs1);
hold on;
plot(t,x,'r:','linewidth',3);
axis([-0.4 0.4 -2 2])
hold off;
xlabel('time');
ylabel('amplitude');
title('under sampling');
legend('fs<2fm');
%critical sampling
fs2=2*f2
ts2=1/fs2;
n2=-0.4:ts2:0.4;
xs2=cos(2.*pi.*f1.*n2)+cos(2.*pi.*f2.*n2);
figure(3);
stem(n2,xs2);
hold on;
plot(t,x,'r:','linewidth',3);
axis([-0.4 0.4 -2 2])
hold off;
xlabel('time');
ylabel('amplitude');
title('critical sampling');
legend('fs=2fm');
%over sampling
fs3=3*f2
ts3=1/fs3;
n3=-0.4:ts3:0.4;
xs3=cos(2.*pi.*f1.*n3)+cos(2.*pi.*f2.*n3);
figure(4);
stem(n3,xs3);
hold on;
plot(t,x,'r:','linewidth',3);
axis([-0.4 0.4 -2 2])
hold off;
xlabel('time');
ylabel('amplitude');
title('over sampling');
legend('fs>2fm');

Result
RESULT:

Hence we verified sampling theorem.


Department of Electronics and Communication Engineering
Analog and Digital Communications Lab

Title of Experiment Verification of PCM


Name of the [Link]
candidate
Register Number 2205a41164
Date of Experiment
Experiment-10

PUSLE CODE MODULATION AND DEMODULATION


AIM
 To study the operation of Pulse Code Modulation & Demodulation by using A.C & D.C
inputs.
 To simulate Pulse Code Modulation and Demodulation.
EQUIPMENT REQUIRED
• PCM modulator trainer
• PCM demodulator trainer
• Storage Oscilloscope
• Digital Multimeter

THEORY
Two basic operations in the conversion of the analog signal into the digital are time
discretization and amplitude discretization. In the context of PCM, the former is accomplished
with the sampling operation and the latter by means of quantization. In addition, PCM involves
another step, namely, conversion of quantized amplitudes into a sequence of simpler pulse
patterns (usually binary), generally called as code words. (The word code in pulse code
modulation refers to the fact that every quantized sample is converted to an R -bit code word.)

Fig. 10.1. A PCM system


Fig. 1.1 illustrates a PCM system. Here, m(t) is the information bearing message signal that is to
be transmitted digitally. m(t) is first sampled and then quantized. The output of the sampler is

m(nTs )  1
t . Ts is the sampling period and n is the appropriate integer. fs  is called
m(t) nTs Ts
the sampling rate or sampling frequency. The quantizer converts each sample to one of the values
that are closest to it from among a pre-selected set of discrete amplitudes. The encoder represents
each one of these quantized samples by an R -bit code word. This bit stream travels on the channel
and reaches the receiving end. With fs as the sampling rate and R -bits per codeword, the bit rate

of the PCM system is R


Rfs  bits/sec. The decoder converts the R -bit code words into the
s T
corresponding (discrete) amplitudes. Finally, the reconstruction filter, acting on these discrete
amplitudes, produces the analog signal, denoted by
mˆ (t) . If there are no channel errors,
mˆ (t)  m(t) . then

Advantages of Pulse Code Modulation (PCM)

 Very High Noise Immunity.


 Due to Digital nature of the signal, repeaters can be placed between the transmitter and
receiver. The repeaters actually regenerate the transmitted PCM Signals. This is not possible in
analog systems. Repeaters further reduce the effect of noise.
 It is possible to store the PCM signal due to its digital nature.
 It is possible to use various coding techniques so that only the desired person can decode the
received signal.
Disadvantages of Pulse Code Modulation (PCM)
• The encoding, decoding and quantizing circuitry of PCM (Pulse Code Modulation) is
Complex.
• Pulse Code Modulation requires a large Bandwidth as compared to the other systems.
Applications
1. In telephony, PCM is used to digitize the Analog Voice (or speech). Digitized voice channel
data rate is 64 kbps.
2. In Space communication, spacecraft transmits signals to earth. Here, the transmitted power is
very low i.e. 10 to 15W and the distances are huge (a few million km). Still, due to the high
noise immunity, only PCM Systems can be used in such applications.

SOFTWARE PROGRAM:
Software Required:
MATLAB R2023b

MATLAB Code:
clc;
clear all;
close all;
a=2;
f=2;
fs=20.*f
t=0:1./fs:1;
% original signal
x=a.*sin(2.*pi.*f.*t);
subplot(2,3,1);
plot(t,x,'r','linewidth',2);
xlabel('time');
ylabel('amplitude');
title('original signal');
grid on
% level shifter
x1=x+a;
subplot(2,3,2);
plot(t,x1,'c','linewidth',2);
xlabel('time');
ylabel('shifted amplitude');
title('level shifted signal');
grid on
% quantization
q=round(x1)
subplot(2,3,3);
plot(t,q,'k','linewidth',2);
xlabel('time');
ylabel('quantized amplitude');
title('quantized signal');
grid on
% Encoder
En=de2bi(q,'left-msb');
subplot(2,3,4);
plot(t,En,'linewidth',2);
xlabel('time');
ylabel('Encoded values');
title('binary bits');
grid on
%decoder
de=bi2de(En,'left-msb');
subplot(2,3,5);
plot(t,de,'c','linewidth',2);
xlabel('time');
ylabel('decoded values');
title('Decimal values');
grid on
% shifting the amplitude level to original amplitude
xr=de-a;
subplot(2,3,6);
plot(t,x,'r-',t,xr,'k+','linewidth',2);
xlabel('time');
ylabel('reconstructed amplitude');
title('reconstructed signal');
grid on

Result

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